This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.
BUG=webrtc:4525
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516993002 .
Cr-Commit-Position: refs/heads/master@{#11192}
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
This allows the test to create its own transports if it, for instance, needs to do demuxing.
BUG=webrtc:5416
Review URL: https://codereview.webrtc.org/1573453002
Cr-Commit-Position: refs/heads/master@{#11187}
To ease use of WebRTC in other codebases, update some macros
to match glibc's ansidecl.h, which uses double-underscores for attributes.
NOTRY=True
Review URL: https://codereview.webrtc.org/1571653002
Cr-Commit-Position: refs/heads/master@{#11185}
Ask the OS for the mic volume every 1 second rather than with every 10
ms chunk. The previous behavior was consuming ~2% of the CPU load of
a voice engine call, and is now negligible.
This is consistent with the webrtc Windows Core Audio implementation,
as well as the Chromium Mac implementation:
https://code.google.com/p/chromium/codesearch#chromium/src/media/audio/agc_audio_stream.h
TEST=voe_cmd_test with AGC continues to work well on Mac.
Review URL: https://codereview.webrtc.org/1564223002
Cr-Commit-Position: refs/heads/master@{#11182}
Tests were failing on android with new libvpx.
vp9 speed setting was changed to 8 recently and some recent changes
in libvpx require update for the tests to pass.
TBR=stefan@webrtc.org
BUG=webrtc:5401
Review URL: https://codereview.webrtc.org/1569903002 .
Cr-Commit-Position: refs/heads/master@{#11173}
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.
Audio streams are using a fake audio device with file input.
The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.
R=pbos@webrtc.orgTBR=kjellander@webrtc.org
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1542653002 .
Cr-Commit-Position: refs/heads/master@{#11171}
Removes multiple index lookups to generated_fec_packets_ speeding up
FecTest.FecTest with >2x in both Debug and Release, improving
performance but also readability.
On Debug this means that the slowest test in modules_tests now takes
~15-20 seconds instead of 50+ seconds, reducing the overall bottleneck.
BUG=webrtc:4712
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1552563003 .
Cr-Commit-Position: refs/heads/master@{#11166}
Update test implementation (test/histograms.h) to be more similar a real implementation (where histogram get functions return a Histogram pointer). Add check that the name of a histogram does not change.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1528403003
Cr-Commit-Position: refs/heads/master@{#11161}
This fixes an assert error in Turnport::OnSendStunPacket
BUG=webrtc:5388
Review URL: https://codereview.webrtc.org/1547373002
Cr-Commit-Position: refs/heads/master@{#11152}
hasn't received anything yet. Deleting an ICE connection before it is pruned or timed out
when it hasn't received anything yet leads to ICE connections being deleted
before they have a chance to send a ping and receive a response.
BUG=
Review URL: https://codereview.webrtc.org/1544003002
Cr-Commit-Position: refs/heads/master@{#11151}
